| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "vie_receiver.h" |
| |
| #include "critical_section_wrapper.h" |
| #include "rtp_dump.h" |
| #include "rtp_rtcp.h" |
| #include "video_coding.h" |
| #include "trace.h" |
| |
| namespace webrtc { |
| |
| ViEReceiver::ViEReceiver(int engine_id, int channel_id, |
| RtpRtcp& rtp_rtcp, |
| VideoCodingModule& module_vcm) |
| : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| engine_id_(engine_id), |
| channel_id_(channel_id), |
| rtp_rtcp_(rtp_rtcp), |
| vcm_(module_vcm), |
| external_decryption_(NULL), |
| decryption_buffer_(NULL), |
| rtp_dump_(NULL), |
| receiving_(false) { |
| } |
| |
| ViEReceiver::~ViEReceiver() { |
| if (decryption_buffer_) { |
| delete[] decryption_buffer_; |
| decryption_buffer_ = NULL; |
| } |
| if (rtp_dump_) { |
| rtp_dump_->Stop(); |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } |
| } |
| |
| int ViEReceiver::RegisterExternalDecryption(Encryption* decryption) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (external_decryption_) { |
| return -1; |
| } |
| decryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu]; |
| if (decryption_buffer_ == NULL) { |
| return -1; |
| } |
| external_decryption_ = decryption; |
| return 0; |
| } |
| |
| int ViEReceiver::DeregisterExternalDecryption() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (external_decryption_ == NULL) { |
| return -1; |
| } |
| external_decryption_ = NULL; |
| return 0; |
| } |
| |
| void ViEReceiver::RegisterSimulcastRtpRtcpModules( |
| const std::list<RtpRtcp*>& rtp_modules) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| rtp_rtcp_simulcast_.clear(); |
| |
| if (!rtp_modules.empty()) { |
| rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(), |
| rtp_modules.begin(), |
| rtp_modules.end()); |
| } |
| } |
| |
| void ViEReceiver::IncomingRTPPacket(const WebRtc_Word8* rtp_packet, |
| const WebRtc_Word32 rtp_packet_length, |
| const WebRtc_Word8* from_ip, |
| const WebRtc_UWord16 from_port) { |
| InsertRTPPacket(rtp_packet, rtp_packet_length); |
| } |
| |
| void ViEReceiver::IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet, |
| const WebRtc_Word32 rtcp_packet_length, |
| const WebRtc_Word8* from_ip, |
| const WebRtc_UWord16 from_port) { |
| InsertRTCPPacket(rtcp_packet, rtcp_packet_length); |
| } |
| |
| int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, |
| int rtp_packet_length) { |
| if (!receiving_) { |
| return -1; |
| } |
| return InsertRTPPacket((const WebRtc_Word8*) rtp_packet, rtp_packet_length); |
| } |
| |
| int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet, |
| int rtcp_packet_length) { |
| if (!receiving_) { |
| return -1; |
| } |
| return InsertRTCPPacket((const WebRtc_Word8*) rtcp_packet, |
| rtcp_packet_length); |
| } |
| |
| WebRtc_Word32 ViEReceiver::OnReceivedPayloadData( |
| const WebRtc_UWord8* payload_data, const WebRtc_UWord16 payload_size, |
| const WebRtcRTPHeader* rtp_header) { |
| if (rtp_header == NULL) { |
| return 0; |
| } |
| |
| if (vcm_.IncomingPacket(payload_data, payload_size, *rtp_header) != 0) { |
| // Check this... |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViEReceiver::InsertRTPPacket(const WebRtc_Word8* rtp_packet, |
| int rtp_packet_length) { |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtp_packet); |
| unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr); |
| int received_packet_length = rtp_packet_length; |
| |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| |
| if (external_decryption_) { |
| int decrypted_length = 0; |
| external_decryption_->decrypt(channel_id_, received_packet, |
| decryption_buffer_, received_packet_length, |
| &decrypted_length); |
| if (decrypted_length <= 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), "RTP decryption failed"); |
| return -1; |
| } else if (decrypted_length > kViEMaxMtu) { |
| WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "InsertRTPPacket: %d bytes is allocated as RTP decrytption" |
| " output, external decryption used %d bytes. => memory is " |
| " now corrupted", kViEMaxMtu, decrypted_length); |
| return -1; |
| } |
| received_packet = decryption_buffer_; |
| received_packet_length = decrypted_length; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(received_packet, |
| static_cast<WebRtc_UWord16>(received_packet_length)); |
| } |
| } |
| return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length); |
| } |
| |
| int ViEReceiver::InsertRTCPPacket(const WebRtc_Word8* rtcp_packet, |
| int rtcp_packet_length) { |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtcp_packet); |
| unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr); |
| int received_packet_length = rtcp_packet_length; |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| |
| if (external_decryption_) { |
| int decrypted_length = 0; |
| external_decryption_->decrypt_rtcp(channel_id_, received_packet, |
| decryption_buffer_, |
| received_packet_length, |
| &decrypted_length); |
| if (decrypted_length <= 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), "RTP decryption failed"); |
| return -1; |
| } else if (decrypted_length > kViEMaxMtu) { |
| WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "InsertRTCPPacket: %d bytes is allocated as RTP " |
| " decrytption output, external decryption used %d bytes. " |
| " => memory is now corrupted", |
| kViEMaxMtu, decrypted_length); |
| return -1; |
| } |
| received_packet = decryption_buffer_; |
| received_packet_length = decrypted_length; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket( |
| received_packet, static_cast<WebRtc_UWord16>(received_packet_length)); |
| } |
| } |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin(); |
| while (it != rtp_rtcp_simulcast_.end()) { |
| RtpRtcp* rtp_rtcp = *it++; |
| rtp_rtcp->IncomingPacket(received_packet, received_packet_length); |
| } |
| } |
| return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length); |
| } |
| |
| void ViEReceiver::StartReceive() { |
| receiving_ = true; |
| } |
| |
| void ViEReceiver::StopReceive() { |
| receiving_ = false; |
| } |
| |
| int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| // Restart it if it already exists and is started |
| rtp_dump_->Stop(); |
| } else { |
| rtp_dump_ = RtpDump::CreateRtpDump(); |
| if (rtp_dump_ == NULL) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StartRTPDump: Failed to create RTP dump"); |
| return -1; |
| } |
| } |
| if (rtp_dump_->Start(file_nameUTF8) != 0) { |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StartRTPDump: Failed to start RTP dump"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViEReceiver::StopRTPDump() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| if (rtp_dump_->IsActive()) { |
| rtp_dump_->Stop(); |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StopRTPDump: Dump not active"); |
| } |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StopRTPDump: RTP dump not started"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| } // namespace webrtc |
